rendered paste body<--- SIP read from 10.1.1.10:5060 --->
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83faf294991fd8abf4b
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 1 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>
Content-Type: application/sdp
Content-Length: 290
v=0
o=Tornado-M20 144768444923955 1 IN IP4 10.1.1.10
s=-
c=IN IP4 10.1.1.10
t=0 0
m=audio 11174 RTP/AVP 18 97 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (11 headers 13 lines) ---
Sending to 10.1.1.10 : 5060 (no NAT)
Using INVITE request as basis request - 25dea83faff8d460fd8abf4a@10.1.1.10
<--- Reliably Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83faf294991fd8abf4b;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>;tag=as03a58c9e
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0d5fd78f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '25dea83faff8d460fd8abf4a@10.1.1.10' in 32000 ms (Method: INVITE)
Found user '100'
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
ACK sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83faf294991fd8abf4b
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>;tag=as03a58c9e
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 1 ACK
User-Agent: Lenoxa.com Tornado-M20
<------------->
--- (8 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83faf297cf1fd8abf48
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 2 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>
Content-Type: application/sdp
Content-Length: 290
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="0d5fd78f",uri="sip:101@10.1.1.7:5060",response="01c09245f4e2200363b21d12f6775311",algorithm=MD5
v=0
o=Tornado-M20 144768444923955 1 IN IP4 10.1.1.10
s=-
c=IN IP4 10.1.1.10
t=0 0
m=audio 11174 RTP/AVP 18 97 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (12 headers 13 lines) ---
Sending to 10.1.1.10 : 5060 (NAT)
Using INVITE request as basis request - 25dea83faff8d460fd8abf4a@10.1.1.10
Found user '100'
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.1.1.10:11174
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Found description format G726-32 for ID 97
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.10:11174
Peer video RTP is at port 10.1.1.10:61184
Looking for 101 in from-internal (domain 10.1.1.7)
list_route: hop: <sip:100@10.1.1.10:5060>
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83faf297cf1fd8abf48;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0
<------------>
Video is at 10.1.1.9 port 13444
Audio is at 10.1.1.9 port 19084
Adding codec 0x100 (g729) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:19:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386
v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19084 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 13444 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83faf297cf1fd8abf48;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>;tag=as3a858e29
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0
<------------>
Retransmitting #1 (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:19:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386
v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19084 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 13444 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
Retransmitting #2 (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:19:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386
v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19084 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 13444 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
Really destroying SIP dialog 'a699a4cd354fcb30fd8abe32@10.1.1.7' Method: REGISTER
Retransmitting #3 (NAT) to 10.1.1.7:5060:
INVITE sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>
Contact: <sip:100@10.1.1.9>
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:19:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 386
v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
b=CT:512
t=0 0
m=audio 19084 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 13444 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
<------------->
--- (8 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 300
v=0
o=Tornado-M20 222889611886663 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 32380 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 5134 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2
<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:32380
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:32380
Peer video RTP is at port 10.1.1.7:5134
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 300
v=0
o=Tornado-M20 222889611886663 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 32380 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 5134 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2
<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:32380
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:32380
Peer video RTP is at port 10.1.1.7:5134
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 300
v=0
o=Tornado-M20 222889611886663 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 32380 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 5134 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2
<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:32380
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:32380
Peer video RTP is at port 10.1.1.7:5134
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 300
v=0
o=Tornado-M20 222889611886663 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 32380 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 5134 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2
<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:32380
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:32380
Peer video RTP is at port 10.1.1.7:5134
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK4ea035d3;rport=5060;received=10.1.1.9
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 INVITE
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>
Content-Type: application/sdp
Content-Length: 300
v=0
o=Tornado-M20 222889611886663 1 IN IP4 10.1.1.7
s=-
c=IN IP4 10.1.1.7
t=0 0
m=audio 32380 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 5134 RTP/AVP 102
a=rtpmap:102 H263-1998/90000
a=fmtp:102 CIF=2;QCIF=2;SQCIF=2
<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 102
Peer audio RTP is at port 10.1.1.7:32380
Found description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Found description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Found description format H263-1998 for ID 102
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x380105 (g723|ulaw|g729|h263|h263p|h264), peer - audio=0x100100 (g729|h263p)/video=0x100000 (h263p), combined - 0x100100 (g729|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.1.7:32380
Peer video RTP is at port 10.1.1.7:5134
list_route: hop: <sip:101@10.1.1.7:5060>
set_destination: Parsing <sip:101@10.1.1.7:5060> for address/port to send to
set_destination: set destination to 10.1.1.7, port 5060
Transmitting (NAT) to 10.1.1.7:5060:
ACK sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK2ada3647;rport
From: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
To: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
Contact: <sip:100@10.1.1.9>
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 10.1.1.9 port 19128
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83faf297cf1fd8abf48;received=10.1.1.10
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>;tag=as3a858e29
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 2351 2351 IN IP4 10.1.1.9
s=session
c=IN IP4 10.1.1.9
t=0 0
m=audio 19128 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
ACK sip:101@10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83fd35eca81fd8abf49
Max-Forwards: 70
From: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
To: <sip:101@10.1.1.7>;tag=as3a858e29
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 2 ACK
User-Agent: Lenoxa.com Tornado-M20
Proxy-Authorization: Digest username="100",realm="asterisk",nonce="0d5fd78f",uri="sip:101@10.1.1.9",response="a88b0f575c73b7cb1a643d72eef6eb96",algorithm=MD5
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '25dea836b6dd54c0fd8abe37@10.1.1.10' Method: REGISTER
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
BYE sip:100@10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f3fbd0f01fd8abc26
Max-Forwards: 70
From: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
To: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 1 BYE
User-Agent: Lenoxa.com Tornado-M20
<------------->
--- (8 headers 0 lines) ---
Sending to 10.1.1.7 : 5060 (NAT)
<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f3fbd0f01fd8abc26;received=10.1.1.7
From: <sip:101@10.1.1.7:5060>;tag=a699a53f37de2593fd8abc39
To: "Pruebas 1" <sip:100@10.1.1.9>;tag=as2d0cf1a2
Call-ID: 4a9006f839abd0c548e9815c484898a4@10.1.1.9
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@10.1.1.9>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '25dea83faff8d460fd8abf4a@10.1.1.10' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:100@10.1.1.10:5060> for address/port to send to
set_destination: set destination to 10.1.1.10, port 5060
Reliably Transmitting (NAT) to 10.1.1.10:5060:
BYE sip:100@10.1.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK0f737fb3;rport
From: <sip:101@10.1.1.7>;tag=as3a858e29
To: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog '4a9006f839abd0c548e9815c484898a4@10.1.1.9' Method: BYE
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK0f737fb3;rport=5060;received=10.1.1.9
From: <sip:101@10.1.1.7>;tag=as3a858e29
To: "100" <sip:100@10.1.1.10>;tag=25dea83fa8012cb2fd8abf4d
Call-ID: 25dea83faff8d460fd8abf4a@10.1.1.10
CSeq: 102 BYE
User-Agent: Lenoxa.com Tornado-M20
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '25dea83faff8d460fd8abf4a@10.1.1.10' Method: ACK
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f24b0b231fd8abc22
Max-Forwards: 70
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 109 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>;expires=60
Expires: 60
Authorization: Digest username="101",realm="asterisk",nonce="11096c4a",uri="sip:10.1.1.9",response="430f1314ca175d5033425f90cbf969c1",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.7 : 5060 (no NAT)
<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f24b0b231fd8abc22;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f24b0b231fd8abc22;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>;tag=as10c2e435
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0bc1b0e3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'a699a4cd354fcb30fd8abe32@10.1.1.7' in 32000 ms (Method: REGISTER)
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f24923d61fd8abc23
Max-Forwards: 70
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 110 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "101" <sip:101@10.1.1.7:5060>;expires=60
Expires: 60
Authorization: Digest username="101",realm="asterisk",nonce="0bc1b0e3",uri="sip:10.1.1.9",response="3c37c5e2a55426d69780f0252674edd2",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.7 : 5060 (NAT)
<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f24923d61fd8abc23;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:101@10.1.1.9>
Content-Length: 0
<------------>
trixbox1*CLI>
<--- Transmitting (NAT) to 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.7:5060;branch=z9hG4bKa699a53f24923d61fd8abc23;received=10.1.1.7
From: "101" <sip:101@10.1.1.9>;tag=a699a4c9b82a08a2fd8abebe
To: "101" <sip:101@10.1.1.9>;tag=as10c2e435
Call-ID: a699a4cd354fcb30fd8abe32@10.1.1.7
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:101@10.1.1.7:5060>;expires=60
Date: Wed, 10 Oct 2007 17:19:51 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'a699a4cd354fcb30fd8abe32@10.1.1.7' in 32000 ms (Method: REGISTER)
Reliably Transmitting (NAT) to 10.1.1.7:5060:
OPTIONS sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK29230794;rport
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as05c7a47c
To: <sip:101@10.1.1.7:5060>
Contact: <sip:Unknown@10.1.1.9>
Call-ID: 3aed28486f7d14a77322a14529ea2952@10.1.1.9
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:19:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK29230794;rport=5060;received=10.1.1.9
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as05c7a47c
To: <sip:101@10.1.1.7:5060>
Call-ID: 3aed28486f7d14a77322a14529ea2952@10.1.1.9
CSeq: 102 OPTIONS
Contact: "101" <sip:101@10.1.1.7:5060>
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '3aed28486f7d14a77322a14529ea2952@10.1.1.9' Method: OPTIONS
Reliably Transmitting (NAT) to 10.1.1.10:5060:
OPTIONS sip:100@10.1.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK2a7982e6;rport
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as0afef62f
To: <sip:100@10.1.1.10:5060>
Contact: <sip:Unknown@10.1.1.9>
Call-ID: 25c13b1a53dde7150e6fd2ee48f56117@10.1.1.9
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 10 Oct 2007 17:19:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK2a7982e6;rport=5060;received=10.1.1.9
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as0afef62f
To: <sip:100@10.1.1.10:5060>
Call-ID: 25c13b1a53dde7150e6fd2ee48f56117@10.1.1.9
CSeq: 102 OPTIONS
Contact: "100" <sip:100@10.1.1.10:5060>
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '25c13b1a53dde7150e6fd2ee48f56117@10.1.1.9' Method: OPTIONS
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83fc03eb401fd8abfb5
Max-Forwards: 70
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 95 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>;expires=60
Expires: 60
Authorization: Digest username="100",realm="asterisk",nonce="50bde35d",uri="sip:10.1.1.9",response="d5fcc516e2a1b3c182b27c381d8d8200",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.10 : 5060 (no NAT)
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83fc03eb401fd8abfb5;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 95 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@10.1.1.9>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83fc03eb401fd8abfb5;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>;tag=as1da93fec
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 95 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b6e3b07"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '25dea836b6dd54c0fd8abe37@10.1.1.10' in 32000 ms (Method: REGISTER)
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
REGISTER sip:10.1.1.9 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83fc0e87691fd8abfb2
Max-Forwards: 70
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 96 REGISTER
User-Agent: Lenoxa.com Tornado-M20
Contact: "100" <sip:100@10.1.1.10:5060>;expires=60
Expires: 60
Authorization: Digest username="100",realm="asterisk",nonce="5b6e3b07",uri="sip:10.1.1.9",response="5d746524af7698565fadd726b12221c7",algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.1.1.10 : 5060 (NAT)
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83fc0e87691fd8abfb2;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 96 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:100@10.1.1.9>
Content-Length: 0
<------------>
trixbox1*CLI>
<--- Transmitting (NAT) to 10.1.1.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK25dea83fc0e87691fd8abfb2;received=10.1.1.10
From: "100" <sip:100@10.1.1.9>;tag=25dea836b6593ed2fd8abe34
To: "100" <sip:100@10.1.1.9>;tag=as1da93fec
Call-ID: 25dea836b6dd54c0fd8abe37@10.1.1.10
CSeq: 96 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:100@10.1.1.10:5060>;expires=60
Date: Wed, 10 Oct 2007 17:19:55 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '25dea836b6dd54c0fd8abe37@10.1.1.10' in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '518b63b96e6e988c38fe9f8572aa704a@10.1.1.9' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 10.1.1.7:5060:
NOTIFY sip:101@10.1.1.7:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK6747f813;rport
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as1d26b37b
To: <sip:101@10.1.1.7:5060>
Contact: <sip:Unknown@10.1.1.9>
Call-ID: 518b63b96e6e988c38fe9f8572aa704a@10.1.1.9
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
ontent-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.1.1.9
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '6d5c54675ffb4c0b0e655855236ad5b1@10.1.1.9' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 10.1.1.10:5060:
NOTIFY sip:100@10.1.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK3b93f1a6;rport
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as03dc8a4b
To: <sip:100@10.1.1.10:5060>
Contact: <sip:Unknown@10.1.1.9>
Call-ID: 6d5c54675ffb4c0b0e655855236ad5b1@10.1.1.9
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.1.1.9
Voice-Message: 0/0 (0/0)
---
trixbox1*CLI>
<--- SIP read from 10.1.1.7:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK6747f813;rport=5060;received=10.1.1.9
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as1d26b37b
To: <sip:101@10.1.1.7:5060>
Call-ID: 518b63b96e6e988c38fe9f8572aa704a@10.1.1.9
CSeq: 102 NOTIFY
<------------->
--- (6 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 10.1.1.10:5060 --->
SIP/2.0 481 Subscription does not exist
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bK3b93f1a6;rport=5060;received=10.1.1.9
From: "Unknown" <sip:Unknown@10.1.1.9>;tag=as03dc8a4b
To: <sip:100@10.1.1.10:5060>
Call-ID: 6d5c54675ffb4c0b0e655855236ad5b1@10.1.1.9
CSeq: 102 NOTIFY
<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '518b63b96e6e988c38fe9f8572aa704a@10.1.1.9' Method: NOTIFY
Really destroying SIP dialog '6d5c54675ffb4c0b0e655855236ad5b1@10.1.1.9' Method: NOTIFY